@Red wrote:

Office A
phone server
Windows server
pfsense firewall

Office B
pfsense firewall
everything else logs into office A using domain credentials

Scenario: Office B is connected to office A across the pfsense vpn. Phones in office B use internal address of phone server, in Office A, to connect the phones.

Issue: Phones in office B have bad quality.

How do i get better quality for phones in Office B since it doesn’t have its own phone server. I can only assume everything is going across the link between offices because it was setup before I started working here. I tried vpn, but doesn’t work because the offices are connected? Ideas?

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@faisalkhan wrote:

Hi all,

I am having strange issue and not able to figure out.

I have Cisco 8861 with 3PCC support and I configured it well from web gui and outbound inbound works well but a strange issue is happening when I dial *97 or voicemail button on the phone it goes on to call pickup with a tone to enter digits but not leading to voicemail box greeting but when I dial *98# it goes to comedian mail perfectly fine.

Not going to voicemail with *97# or voicemail button.

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@Vinnie881 wrote:

We have answering machine detection (AMD_APP) running and on our dialer roughly 1 out of every 200 or so calls has AMD return HANGUP, but it’s after 300 seconds? The 300 seconds looks like a rtp timeout, but I set under sip settings RTP Timeout and RTP HOLD TImeout to 60 seconds.

Any ideas what can be going on here? We are using an EC2 aws instance for our pbx, but like i said 99%+ of our calls go out no issue, it’s just very random. I verified our firewall ports were allowing ports 10000-20000 udp on both the local box and the aws firewall…

Worst case scenerio how do I have my amd just timeout on it’s own if it is in the app too long? Here are my current settings

Initial Silence
2500
Greeting
1500
After Greeting Silence
800
Total Analysis Time
5000
Minimum Word Length
100
Maximum Word Length
5000
Between Words Silence
50
Maximum Number Of Words
3
Silence Threshold
256

Any idea how to get AMD to auto proceed so it does not get stuck in some weird limbo in this scenerio, Or better fix this issue completely?

Thanks!

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@CReynolds wrote:

I am looking for any information with regard to if PJSIP support of the text codec t140 and t140red.

chan_sip supports this now but I cannot find any information and have had no luck implementing t140 with PJSIP. Does anyone know if its currently supported or if there are any efforts planned or underway to implement support?

Thank you all for any help or information!

core show codecs
39 text red red (T.140 Realtime Text with redundancy)
40 text t140 t140 (Passthrough T.140 Realtime Text)

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@mgbolts wrote:

Hi,

Hope I am not posing a common question…I did have a good hunt around for a solution. I am running FreePBX 14 behind a pfsense firewall. I have a dedicated static external IP and Internal IPs which are Natted 1:1.

I have followed a couple of guides which improved the time from 15mins to 60mins but it still drops …I changed the firewall optimisation setting to conservative and also Disabled the Firewall Scrub.

I am guessing its still the firewall…keen to hear any ideas as what to try next.

Log for call drop is detailed below (note: My number has been starred out)

-- Channel SIP/Aus_Phone_Co-00000005 left 'simple_bridge' basic-bridge <f0b079b3-7c8e-4bc9-8d13-8b24f5df1300>
-- Channel SIP/210-00000004 left 'simple_bridge' basic-bridge <f0b079b3-7c8e-4bc9-8d13-8b24f5df1300>

== Spawn extension (macro-dialout-trunk, s, 32) exited non-zero on ‘SIP/210-00000004’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, **********, 6) exited non-zero on ‘SIP/210-00000004’
– Executing [h@from-internal:1] Macro(“SIP/210-00000004”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/210-00000004”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/210-00000004”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“SIP/210-00000004”, "SIP/Aus_Phone_Co-00000005 monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“SIP/210-00000004”, “attendedtransfer-rec-restart.php,SIP/Aus_Phone_Co-00000005,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <SIP/210-00000004>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“SIP/210-00000004”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/210-00000004’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/210-00000004’
– SIP/210-00000004 Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“SIP/210-00000004”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“SIP/210-00000004”, “HANGUP CAUSE: 16”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“SIP/210-00000004”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“SIP/210-00000004”, “MASTER CHANNEL: 1547877720.4 = 1547877720.4”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“SIP/210-00000004”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“SIP/210-00000004”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“SIP/210-00000004”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <SIP/210-00000004>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“SIP/210-00000004”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/210-00000004’
– SIP/210-00000004 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

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@captain118 wrote:

Our vulnerability scanner has reported several issues with the web server’s configuration in Freepbx 14.
I have no problem with correcting the issues however all the config files are autogenerated how do I ensure that my changes wont be reverted in the next upgrade?

The primary issues detected are regarding weak ciphers being offered, httpOnly cookie not set and clear text passwords for port 81.

Also installing mod_security would be a good touch. Has anyone done that without any negative side affects?

Thanks,
Kirk

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@nexet wrote:

hi guys, I can not find the configuration files of my trunks, I can not access via http anymore, I should save the trunks, the output and entry routes, where can I find the files?
I’m looking in / etc / asterisk but I can not find them

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@dlemmink wrote:

Moved my installation to new hardware with FreePBX 14.0.5.25.

I am no longer seeing the console debug messages I saw previously when debugging agi scripts.
Have tried debug, verbose and agi setting and still am not receiving the console debug messages.

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@danotec2009 wrote:

New here… hoping to get some insight on what I hope will help my situation.

I am the IT administrator for a small tech company. We offer managed services and 24/7 support. Outside of business hours we have a rotating oncall schedule that consists of 1 primary technician, 1 backup technician, and 1 dept. manager, with oncall priority in that order.

For our phone system we are provisioned to a hosted provider (Voyant). We subscribe to 25 premium seats, and 4 standard seats. We use the 4 standard seats with 4 ATA’s to facilitate our legacy oncall voicemail system, a Panasonic TVA-50 voicemail appliance.

The reason we continue to use the legacy voicemail box, vs. other solutions, is because we cannot find a solution that offers us the same features that we are dependent on.

Here is how the system currently works, and what we are trying to mirror in another solution:

Customer calls the hunt group phone number that is registered to all 4 ATA’s. The voicemail box then answers the call with our emergency voicemail greeting, instructing the customer to leave a message and wait for a call back. The voicemail box then calls the primary oncall technician, if the technician answers, the voicemail box give them the option to take and listen to the message. If the technican doesn’t answer, and/or the message is not listened to within 5 minutes, the system then calls the backup technician, and the cycle starts again finally ending with a call to the manager. In the GUI of the voicemail box, you can specify who is 1st, 2nd, and 3rd in the calling order, and we manually change this twice a week when the schedule rotates.

Is it possible to accomplish this using Freebpx? Here is what I’d like to make happen:

1. Host the Freepbx instance in a VM
2. Use a SIP trunk as the primary inbound/outbound calling method, which would allow us to eliminate the 4 ATA’s and the associated service fees.
3. Mirror the functionality our current system using an add-on like VM Notify (can’t post a link).

Based on the above, I’m unsure of a few things:

1. I know I can run freepbx in a VM, but does the VM need to make any external connections other than network?
2. Does 1 SIP Trunk cover several concurrent inbound/outbound calls at once? This would need to replace the concurrent capacity of 4 analog lines.

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@pasi wrote:

Does anyone know an easy way to disable the auto update check of the MicroSIP application?

I was looking into the microsip.ini in my C:\Users%username%\AppData\Roaming\MicroSIP and found 2 variables of

updatesInterval=
checkUpdatesTime=1547947176

I’m not sure if changing these values might help me archive what I’m looking for.

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@luckman212 wrote:

I’ve got a click-to-dial (CTI) script written in PHP that sits on my FreePBX server, and uses AMI to initiate the outbound call and bridge it to the local extension. It works fine!

The problem I’m trying to solve is, since Asterisk is making 2 call legs, the phone on my desk displays From: Mary when I am actually calling To: Mary.

Another unwanted side effect is that the call doesn’t register as an outbound call, so “redial” doesn’t work as expected, etc.

Is there any SIP header I can add that will update the call display & direction to be an OUTBOUND call and show “To:XXXXX” instead?

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@gdesilva wrote:

Hi, I am running Raspbx 14.0.5.25 with recently upgraded voicemail module (14.0.4.2).

Before updating the voicemail module, the MWI light would automatically go off after I delete all voice mail messages and hang up the receivers on my handsets.

But now after I delete the voicemail messages and hangup the handset the MWI light stays on till the handset issues a new SIP Registration Request.

Has anyone come across this issue? It is not a critical problem but would like to know whether there is any setting I need to tweak to get MWI working properly.

Many thanks.

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@jorgeraggs wrote:

Hi All,

My first post. I have a problem with TLS on PJSIP (not using SIP at the moment), I was trying to fix it in the last few days but with no luck, so decided to ask for help here. I have some experience with VoIP and encryption, not on FreePBX/Asterisk though.

My configuration:
PJSIP udp/5060, tcp/5062, tls/5065
firewall ports open (telnet to 5060, no NAT (VM in the cloud with public IP), lsof command shows
COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME
asterisk 1180 asterisk 43u IPv4 18309 0t0 UDP *:sip
asterisk 1180 asterisk 16u IPv4 18810 0t0 TCP *:5065 (LISTEN)

I removed the default certificate authority and generated it from scratch, together a new self-signed certificate and respective pkcs12/pem for the client. I uploaded CA and pkcs12 to the client.

Problem however is that in the tshark I can see that TLS fails. I even tried to change it to lower version, but it fails anyway.
7 3.289519851 ext_home_IP ext_ast_IP TLSv1 374 Client Hello
8 3.289537015 ext_ast_IP ext_home_IP TCP 56 5065 54101 [ACK] Seq=1 Ack=319 Win=30336 Len=0
9 3.298714344 ext_ast_IP ext_home_IP TLSv1.2 63 Alert (Level: Fatal, Description: Handshake Failure)
10 3.298811682 ext_ast_IP ext_home_IP TCP 56 5065 54101 [FIN, ACK] Seq=8 Ack=319 Win=30336 Len=0
11 3.329211013 ext_home_IP ext_ast_IP TCP 56 54101 5065 [ACK] Seq=319 Ack=8 Win=17408 Len=0

I tested this with both IP address and FQDN for the registrar server (using the same FQDN for the certificate authority and respective certificates). I still get the same error and of course can’t make a call saying that the

[2019-01-19 15:18:41] ERROR[20726]: res_pjsip/config_transport.c:665 transport_apply: Transport ‘0.0.0.0-tcp’ could not be started: Address already in use
[2019-01-19 15:18:41] ERROR[20726]: res_sorcery_config.c:407 sorcery_config_internal_load: Could not create an object of type ‘transport’ with id ‘0.0.0.0-tcp’ from configuration file ‘pjsip.conf’

Now I’m wondering how these two things can be related? Do I need to have TCP and TLS on the same port? But I’m using DTLS, shouldn’t that use UDP? I’ve seen some older post where there was some change required on the extension, but haven’t found it in new FreePBX (autodomain since I was trying to use FQDN and few other flags)

I’ve read through the community couple of times, tried this and that, but I’m not sure what else can I test to make it working. When I revert back to UDP, all I can make calls fine again, with two-way audio.

Any advise is appreciated,
Kind regards,

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@jimbo108 wrote:

Hi All, first time post and relatively new to FreePBX. I have installed an EXP40-1 expansion module on a Yealink T46G phone and can manually configure this through the web interface. I would like to get this to apply the template that is set up on the FreePBX web admin portal. So far I have set “Set Module Admin to Edge mode” to “Yes” and ensured the device is selected in extension mapping.

Thanks in advance, Jim.

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@marektom wrote:

Hello,
Please help with the HT503 configuration problem as a trunk for FreePBX. I’ve tried a lot of instructions. I attach the settings pictures on google drive. I do not even have incoming or outgoing calls.
IP HT503: 10.181.28.89
IP FreePBX: 10.181.28.91
Please help. I’m really desperate.
thank you very much

I can not upload pictures or links, so please if anyone wants to help me contact me and send pictures to him.
email: tomas.marek(at)agentes-it(dot)com

Snímek obrazovky 2019-01-21 v 12.46.51.png1420x677 116 KB

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@cdsJerryw wrote:

One of our Sangoma phones vanished from Extension Mapping in the middle of the day today. I’ve been unable to get it back. There’s nothing showing on the firewall as blocked. As near as I can tell, Fail2ban isn’t blocking it. Nothing was changed on the FreePBX prior it it’s loss (since last week when I did a module update but the phone worked after that).

I can ping the phone from the remote router. I can ping the FreePBX IP from the remote router. But the S500 phone will not connect to FreePBX. I’ve been trying for hours now. I’ve rebooted everything, including FreePBX. No success.

FreePBX 14.0.5.25 Asterisk 13.22.0

What’s the next step in troubleshooting this problem? I’m not sure where to turn next.

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@jdrichmond wrote:

Hi I am hoping someone can point in the right direction here. We are running Freepbx and it has been working perfect for years now.

We had an 8 port analog card at first and just switch to (not sure what you call it) T1? Let me explain. The phone company installed fiber and with that all phone lines are now going over that to the special box they installed. Then from there we ran a network cable from there box to a second network port of the freepbx and got that all working.

The only issue is when we lose internet we also lose all incoming call capabilities. Outbound works fine with no internet.
Incoming/outgoing calls are working by a sip trunk. Here are those settings

type=peer
qualify=no
nat=yes
insecure=very
host=10.0.0.1
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw

Not sure if you need any logs or what logs you would want. If you tell me what to provide I would be more than happy to do so.

Thank you in advance for the help!

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@bajramia wrote:

Hi All,
I have a client who have PBXact UC 60, he is having long delays when he calls outbound, and when he call internal which all internal extension are set auto-answer, it is about 5 second delay before the call gose through. Also somtimes his calls drops, he is also complaining about the voice is not clear.
He only have 13 extensions so the load is not there.

Any ideas.

Thank you

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@iceget wrote:

dear community,

i have a little problem in my freepbx.

i have created for my phones a queue.
so if i now on the phone and a new call comes in (and is in the queue),
after i hang up my current call, i cannot get the call or my phone is not ringing again, …
i cannot get the call from my phone.

can anybody helps me how to setup that directly if i have done the hang up,
that the callers in queue should at the same moment ring too on my phone?
where is this setting? or what i have do configure?

thank you very much.

many greets

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@jorgeraggs wrote:

Hi,

I’m having a problem with the CDRs not being written into the database lately. They have never worked, but I think I’m rather close to fix this. I was hoping someone can help understanding what the problem is. I’m running Asterisk 16 and FreePBX 15 on Debian.

I can connect to database with mysql and the below configured user fine. I cannot somehow make connection between asterisk and database and I get this error

– Reloading module ‘res_odbc.so’ (ODBC resource)
[2019-01-22 16:20:24] WARNING[7978]: res_odbc.c:941 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified
[2019-01-22 16:20:24] NOTICE[7978]: res_odbc.c:596 load_odbc_config: Registered ODBC class ‘asteriskcdrdb’ dsn->[MySQL-asteriskcdrdb]
[2019-01-22 16:20:24] NOTICE[7978]: res_odbc.c:488 load_odbc_config: Adding ENV var: ODBCSYSINI=/etc
[2019-01-22 16:20:24] NOTICE[7978]: res_odbc.c:488 load_odbc_config: Adding ENV var: ODBCINI=/etc/odbc.ini
– Reloading module ‘extconfig’ (Configuration)
– Reloading module ‘res_config_odbc.so’ (Realtime ODBC configuration)
– Reloading module ‘cdr’ (CDR Engine)
[2019-01-22 16:20:24] NOTICE[7978]: cdr.c:4485 cdr_toggle_runtime_options: CDR simple logging enabled.
– Reloading module ‘cdr_adaptive_odbc.so’ (Adaptive ODBC CDR backend)
[2019-01-22 16:20:24] WARNING[7978]: res_odbc.c:941 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified
[2019-01-22 16:20:24] WARNING[7978]: cdr_adaptive_odbc.c:136 load_config: No such connection ‘asteriskcdrdb’ in the ‘asteriskcdrdb’ section of cdr_adaptive_odbc.conf. Check res_odbc.conf.

<odbc.ini>
[asteriskcdrdb]
Description=MySQL connection to ‘asteriskcdrdb’ database
#driver=MySQL
driver=/usr/lib/x86_64-linux-gnu/libmysqlclient.so
server=localhost
database=asteriskcdrdb
Port=3306
Socket=/var/run/mysqld/mysql.sock
option=3
Charset=utf8
Trace=Yes
username=cdruser
password=password

<odbcinst.ini>
[MySQL]
Description = ODBC for MySQL
Driver64 = /usr/lib/x86_64-linux-gnu/libodbc.so
Setup64 = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so
#Driver64 = /usr/lib64/libmyodbc5.so
#Setup64 = /usr/lib64/libodbcmyS.so
FileUsage = 1
Pooling = Yes
CPTimeout = 120

<res_odbc.conf>
#include res_odbc_custom.conf
#include res_odbc_additional.conf
[ENV]
ODBCSYSINI => /etc
ODBCINI => /etc/odbc.ini

<res_odbc_additional.conf>
[asteriskcdrdb]
enabled=>yes
dsn=>MySQL-asteriskcdrdb
pre-connect=>yes
max_connections=>5
username=>cdruser
password=>password
database=>asteriskcdrdb

Now I think that error tries to tell me I don’t have Data Source configured in the odbc.ini, but I don’t know how to configure it. I’ve found this somewhere else, but I’m struggling to see what should I use for Asterisk

[ODBC Data Sources]
MapR Drill 32-bit=MapR Drill ODBC Driver 32-bit
MapR Drill 64-bit=MapR Drill ODBC Driver 64-bit

Secondly, even isql is giving me error.

isql -v asteriskcdrdb cdruser password
[IM004][unixODBC][Driver Manager]Driver’s SQLAllocHandle on SQL_HANDLE_HENV failed
[ISQL]ERROR: Could not SQLConnect

Does anyone please know how to configure Data Source in odbc configuration and how can I perhaps make changes to the configuration in FreePBX GUI (as I cannot make changes to the res_odbc.conf in shell) so I can proceed? I’m out of ideas.

Thank you,
Kind regards,

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